diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.cc | 77 |
1 files changed, 77 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.cc b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.cc new file mode 100644 index 0000000000..41efdad2c8 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.cc @@ -0,0 +1,77 @@ +/* + * Copyright (c) 2021 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_processing/agc2/saturation_protector_buffer.h" + +#include "rtc_base/checks.h" +#include "rtc_base/numerics/safe_compare.h" + +namespace webrtc { + +SaturationProtectorBuffer::SaturationProtectorBuffer() = default; + +SaturationProtectorBuffer::~SaturationProtectorBuffer() = default; + +bool SaturationProtectorBuffer::operator==( + const SaturationProtectorBuffer& b) const { + RTC_DCHECK_LE(size_, buffer_.size()); + RTC_DCHECK_LE(b.size_, b.buffer_.size()); + if (size_ != b.size_) { + return false; + } + for (int i = 0, i0 = FrontIndex(), i1 = b.FrontIndex(); i < size_; + ++i, ++i0, ++i1) { + if (buffer_[i0 % buffer_.size()] != b.buffer_[i1 % b.buffer_.size()]) { + return false; + } + } + return true; +} + +int SaturationProtectorBuffer::Capacity() const { + return buffer_.size(); +} + +int SaturationProtectorBuffer::Size() const { + return size_; +} + +void SaturationProtectorBuffer::Reset() { + next_ = 0; + size_ = 0; +} + +void SaturationProtectorBuffer::PushBack(float v) { + RTC_DCHECK_GE(next_, 0); + RTC_DCHECK_GE(size_, 0); + RTC_DCHECK_LT(next_, buffer_.size()); + RTC_DCHECK_LE(size_, buffer_.size()); + buffer_[next_++] = v; + if (rtc::SafeEq(next_, buffer_.size())) { + next_ = 0; + } + if (rtc::SafeLt(size_, buffer_.size())) { + size_++; + } +} + +absl::optional<float> SaturationProtectorBuffer::Front() const { + if (size_ == 0) { + return absl::nullopt; + } + RTC_DCHECK_LT(FrontIndex(), buffer_.size()); + return buffer_[FrontIndex()]; +} + +int SaturationProtectorBuffer::FrontIndex() const { + return rtc::SafeEq(size_, buffer_.size()) ? next_ : 0; +} + +} // namespace webrtc |